blob: 27df7e5a0837e9fa3d3d3c860cb0a586d779df6e [file] [log] [blame]
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
* FEC and NACK added bitrate is handled outside class
#include <deque>
#include <utility>
#include <vector>
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class RtcEventLog;
class SendSideBandwidthEstimation {
SendSideBandwidthEstimation() = delete;
explicit SendSideBandwidthEstimation(RtcEventLog* event_log);
virtual ~SendSideBandwidthEstimation();
void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
// Call periodically to update estimate.
void UpdateEstimate(int64_t now_ms);
// Call when we receive a RTCP message with TMMBR or REMB.
void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth);
// Call when a new delay-based estimate is available.
void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps);
// Call when we receive a RTCP message with a ReceiveBlock.
void UpdateReceiverBlock(uint8_t fraction_loss,
int64_t rtt,
int number_of_packets,
int64_t now_ms);
void SetBitrates(int send_bitrate,
int min_bitrate,
int max_bitrate);
void SetSendBitrate(int bitrate);
void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
int GetMinBitrate() const;
enum UmaState { kNoUpdate, kFirstDone, kDone };
bool IsInStartPhase(int64_t now_ms) const;
void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
// Updates history of min bitrates.
// After this method returns min_bitrate_history_.front().second contains the
// min bitrate used during last kBweIncreaseIntervalMs.
void UpdateMinHistory(int64_t now_ms);
// Cap |bitrate_bps| to [min_bitrate_configured_, max_bitrate_configured_] and
// set |current_bitrate_bps_| to the capped value and updates the event log.
void CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate_bps);
std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
// incoming filters
int lost_packets_since_last_loss_update_Q8_;
int expected_packets_since_last_loss_update_;
uint32_t current_bitrate_bps_;
uint32_t min_bitrate_configured_;
uint32_t max_bitrate_configured_;
int64_t last_low_bitrate_log_ms_;
bool has_decreased_since_last_fraction_loss_;
int64_t last_feedback_ms_;
int64_t last_packet_report_ms_;
int64_t last_timeout_ms_;
uint8_t last_fraction_loss_;
uint8_t last_logged_fraction_loss_;
int64_t last_round_trip_time_ms_;
uint32_t bwe_incoming_;
uint32_t delay_based_bitrate_bps_;
int64_t time_last_decrease_ms_;
int64_t first_report_time_ms_;
int initially_lost_packets_;
int bitrate_at_2_seconds_kbps_;
UmaState uma_update_state_;
std::vector<bool> rampup_uma_stats_updated_;
RtcEventLog* event_log_;
int64_t last_rtc_event_log_ms_;
bool in_timeout_experiment_;
float low_loss_threshold_;
float high_loss_threshold_;
uint32_t bitrate_threshold_bps_;
} // namespace webrtc